similar to: error: CDR on channel '<unknown>' has not started

Hi all ! I'm trying to make a automonitor generated filename to "make its way" into CRD(usrefiled), so I can keep track of recorded conversations in CDR logs. Looking how to do that, I have found cool (but almost undocumented) option of res_monitor: if you set monitor format in form of "format:<string>" (i.e. "wav:monitor"), res_monitor will prefix the

Hi. Asterisk 1.4.12.1. I get a lot of message like this. Someone knows what this message mean? Do i have to worry about it? [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Local/152 at local-f137,1' not posted [Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on channel 'Agent/152' not posted [Nov 7 15:24:25] NOTICE[31247]: cdr.c:434

Hey gang, Currently I have this dialplan: exten => _9.,1,Dial(blah/blah) exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DeadAGI(rate_call.php) The AGI script takes the completed call, determines all that NPNAXX crap, finds the cost and then updates the CDR with the cost. Problem is, I keep getting these messages: Jan 4 13:25:36 WARNING[13689]: cdr.c:114

Every once in a while (like 2 out of 7 times), I get the following message: [Nov 14 12:49:02] NOTICE[6855]: cdr.c:434 ast_cdr_free: CDR on channel 'SIP/5000-082508f0' not posted I look in the cdr table in mySQL and indeed, the record is not posted for that call. This makes me want to create hard file and a compare script between the file cdr and the odbc cdr, but I was wondering if

Hi, When I use NoCDR application I obtain this warning in console log: Mar 1 11:16:08 WARNING[3513]: cdr.c:114 ast_cdr_free: CDR on channel 'SIP/492-7371' not posted Mar 1 11:16:08 WARNING[3513]: cdr.c:116 ast_cdr_free: CDR on channel 'SIP/492-7371' lacks end Can someone explain to me what is due? Thanks.

I have been having some troubles with the zaptel channel on what appears to be the inbound process. The box is running the stable CVS code and has a TDM400P 4-port FXO card in it for analog connectivity. Channel 1 is the only active port on the card at the moment as we only have one analog line. What has been happening is that it looks like Asterisk has been detecting an inbound call even though

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>

Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was

hello, I'm getting a lot of messages in the log file like : May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/1913@from-internal-a363,2' not posted May 2 17:12:07 NOTICE[15877] cdr.c: CDR on channel ' Local/1913@from-internal-a363,2' lacks end May 2 17:12:07 WARNING[15901] func_db.c: DB requires an argument, DB(<family>/<key>) May 2 17:12:07

Hi all, I'm having a problem with an asterisk server being unable to call certain cellphones and answering machines. Anytime the person answers the phone call, everything works well. But when the call goes to voicemail or an answering machine, I get the error message below: ===================================== *CLI> -- Attempting call on IAX2/u92613106/15551234567 for

I have an Asterisk server (1.4.13) using PRI in Monterrey, Mexico. This is really the first server I have used with PRI in Mexico as we normally use MFC/R2. Everything seems to be working except that some numbers always seem to be busy when you dial them. All these numbers belong to different phone companies. I know that with R2 this problem is present if you have a "#define

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

Hello! I have tried to compile the capi module (http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz) on fedora2 today. ----------- MAKEFILE -------------- ..EXPORT_ALL_VARIABLES: INSTALL_PREFIX= #ASTERISK_HEADER_DIR=$(INSTALL_PREFIX)/usr/include ASTERISK_HEADER_DIR=/usr/include/asterisk #MODULES_DIR=$(INSTALL_PREFIX)/usr/lib/asterisk/modules

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work fine. To avoid problems with the phone line (german E1) I called the provider, he did a 45 min. route

I just managed to put in a TE410 card in an Asterisk box to work with OnRamp 20(E1 downunder). I am able to dial in but was not able to dial out. Can anyone offer me some advice please? In my extensions.conf, I just put in: [default] ... exten => 0,1,Dial(Zap/g1) and I get this on the console when I dialled 0. -- Executing [0 at default:1] Dial("SIP/5166-b76004f8",

hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [450 at lacnicuy:4] GotoIf("Zap/31-1", "0?6:5") in new stack -- Goto (lacnicuy,450,5) -- Executing [450 at lacnicuy:5] Dial("Zap/31-1", "IAX2/lacnic:splacnic at

Now on to another problem that we've had as far as I know since the beginning of using Asterisk 9+ months ago. I've been trying very hard to knock this problem out but regardless of what I do, it's still there. So, the problem is, when a call is in the parking lot, it then times out after whatever time frame and dials the extension that put it on hold. After 60 seconds of ringing

Please help with this strange issue. When "sip show peers" returns status "Unknown" the CDR does not include the accountcode even though the call is correctly processed. I'm using A2 Billing and it uses the accountcode to determine the authentication. Asterisk version 1.4.21.2 I'm calling from a Quintum device. I'm very puzzeled. Name/username Host

I encountered this strange problem which is I can call into Asterisk box but I cannot call out. I was successful before using exactly the same euroISDN line but with TE110 and different versions of Asterisk. This time, I am using: . TE410 . Asterisk 1.4.13 . Zaptel 1.4.6 . Libpri 1.4.2 1) I put the following into extensions.conf to get to the outside line exten => 0,1,Dial(Zap/1) 2)

Hi guys, I would llike to ask for your help on a problem I'm having with the cdr functionality. I installed asterisk 1.0.4, and asterisk-addons-1.0.4 and followed the procedures for installation and mysql configuration. Everything seems fina. The cdr_mysql module is loaded, and I get no error messages. But the cdr's are not going to the mysql database, they are going to a csv file in